• Title/Summary/Keyword: speech source

Search Result 281, Processing Time 0.022 seconds

Learners' Sociolinguistic Behavior: In Search of Four Major Sources of Pragmatic Errors

  • Suh, Jae-Suk
    • English Language & Literature Teaching
    • /
    • v.7 no.1
    • /
    • pp.35-48
    • /
    • 2001
  • One of the areas of second language acquisition that enjoyed popularity in recent years is interlanguage pragmatics. The main reason for this popularity lies in the critical role of pragmatic competence in appropriate use of a target language. The aim of this paper was to examine L2 learners' pragmatic behavior in their speech act performance and determine main sources causing pragmatic difficulty. Four major sources of pragmatic errors were identified: linguistic proficiency, L1 transfer, waffling and teaching activities. Each source was discussed with empirical evidence in some detail, and teaching suggestions were provided for developing learners' pragmatic competence in EFL classrooms.

  • PDF

Application of AMDF for Improvement of algorithm in estimation sytem of speech source (음원위치 추정 시스템에서 속도향상을 위한 AMDF의 적용)

  • 송도훈
    • Proceedings of the Acoustical Society of Korea Conference
    • /
    • 1998.06d
    • /
    • pp.64-67
    • /
    • 1998
  • 원격지간 화상회의 시스템에서 화자의 위치에 따른 카메라 제어를 위해서는 마이크로폰 배렬(Microphone Array)로 수음한 음성신호에 대해 각 마이크로폰간의 빠른 지연시간 추정이 요구된다. 본 연구에서는 음원위치 추정을 위한 지연시간(Time delay) 계산을 위해 AMDF(Average Magnitude Difference Function)를 적용하여 연산시간을 단축시키는데 목적을 두고 있다. 기본의 상호상관함수 (Cross-correlation )알고리즘 과 본 연구에서 적용한 AMDF 알고리즘을 비교하기 위해 SNR 10dB 와 20dB 인 200Hz, 500Hz, 1kHz, 2kHz의 정현파 합성신호와 단음절 음성신호에 대해 시뮬레이션을 행하였다. 시뮬레이션 결과 AMDF 알고리즘의 정확한 지연시간 추정을 확인하였다.

  • PDF

Frequency Domain Blind Source Seperation Using Cross-Correlation of Input Signals (입력신호 상호상관을 이용한 주파수 영역 블라인드 음원 분리)

  • Sung Chang Sook;Park Jang Sik;Son Kyung Sik;Park Keun-Soo
    • Journal of Korea Multimedia Society
    • /
    • v.8 no.3
    • /
    • pp.328-335
    • /
    • 2005
  • This paper proposes a frequency domain independent component analysis (ICA) algorithm to separate the mixed speech signals using a multiple microphone array By estimating the delay timings using a input cross-correlation, even in the delayed mixture case, we propose a good initial value setting method which leads to optimal convergence. To reduce the calculation, separation process is performed at frequency domain. The results of simulations confirms the better performances of the proposed algorithm.

  • PDF

An efficient method of spatial cues and compensation method of spectrums on multichannel spatial audio coding (멀티채널 Spatial Audio Coding에서의 효율적인 Spatial Cues 사용과 그에 따른 Spectrum 보상방법)

  • Lee, Byong-Hwa;Beack, Seung-Kwon;Seo, Jeong-Gil;Han, Min-Soo
    • MALSORI
    • /
    • no.53
    • /
    • pp.157-169
    • /
    • 2005
  • This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

  • PDF

A Phonetic Study of 'Sasang Constitution' (음성학적으로 본 사상체질)

  • Moon, Seung-Jae;Tak, Ji-Hyun;Hwang, Hye-Jeong
    • Proceedings of the KSPS conference
    • /
    • 2005.04a
    • /
    • pp.63-66
    • /
    • 2005
  • Sasang Constitution, one branch of oriental medicine, claims that people can be classified into four different 'constitutions:' Taeyang, Taeum, Soyang, and Soeum. This study investigates whether the classification of the 'constitutions' could be accurately made solely based on people's voice by analyzing the data from 46 different voices whose constitutions were already determined. Seven source-related parameters and four filter-related parameters were phonetically analyzed and the GMM(gaussian mixture model) was tried with the data. Both the results from phonetic analyses and GMM showed that all the parameters (except one)failed to distinguish the constitutions of the people successfully. And even the single exception, the bandwidth of F2, did not provide us with sufficient reasons to be the source of distinction. This result seems to suggest one of the two conclusions: either the Sasang Constitutions cannot be substantiated with phonetic characteristics of peoples' voices with reliable accuracy, or we need to find yet some other parameters which haven't been conventionally proposed.

  • PDF

Source Localization Based on Independent Doublet Array (독립적인 센서쌍 배열에 기반한 음원 위치추정 기법)

  • Choi, Young Doo;Lee, Ho Jin;Yoon, Kyung Sik;Lee, Kyun Kyung
    • Journal of the Institute of Electronics and Information Engineers
    • /
    • v.51 no.10
    • /
    • pp.164-170
    • /
    • 2014
  • A single near-field sounde source bearing and ranging method based on a independent doublet array is proposed. In the common case of bearing estimation method, unform linear array or uniform circular array are used. It is constrained retaining aperture because of array structure to estimate the distance of the sound source. Recent using independent doublet array sound source's bearing and distance esmtimation method is proposed by wide aperture. It is limited to the case doublets are located on a straight line. In this paper, we generalize the case and estimate the localization of a sound source in the various array structure. The proposed algorithm was verified performance through simulation.

An efficient space dividing method for the two-dimensional sound source localization (2차원 상의 음원위치 추정을 위한 효율적인 영역분할방법)

  • Kim, Hwan-Yong;Choi, Hong-Sub
    • The Journal of the Acoustical Society of Korea
    • /
    • v.35 no.5
    • /
    • pp.358-367
    • /
    • 2016
  • SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

Robust Blind Source Separation to Noisy Environment For Speech Recognition in Car (차량용 음성인식을 위한 주변잡음에 강건한 브라인드 음원분리)

  • Kim, Hyun-Tae;Park, Jang-Sik
    • The Journal of the Korea Contents Association
    • /
    • v.6 no.12
    • /
    • pp.89-95
    • /
    • 2006
  • The performance of blind source separation(BSS) using independent component analysis (ICA) declines significantly in a reverberant environment. A post-processing method proposed in this paper was designed to remove the residual component precisely. The proposed method used modified NLMS(normalized least mean square) filter in frequency domain, to estimate cross-talk path that causes residual cross-talk components. Residual cross-talk components in one channel is correspond to direct components in another channel. Therefore, we can estimate cross-talk path using another channel input signals from adaptive filter. Step size is normalized by input signal power in conventional NLMS filter, but it is normalized by sum of input signal power and error signal power in modified NLMS filter. By using this method, we can prevent misadjustment of filter weights. The estimated residual cross-talk components are subtracted by non-stationary spectral subtraction. The computer simulation results using speech signals show that the proposed method improves the noise reduction ratio(NRR) by approximately 3dB on conventional FDICA.

  • PDF

A Study on the Acoustical Characteristics of Pistol Impluse and MLS Source Measurements in Room Types (음향측정시 실의 종류와 음원에 따르는 음향인자 측정분석에 관한 연구)

  • Kim, Jeong-Jung;Son, Jang-Ryeol
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
    • /
    • 2004.11a
    • /
    • pp.1028-1031
    • /
    • 2004
  • Last target of architectural acoustics is that people wish to convey voice effectively from the space adaptively in use purpose in building. But, to how exactly through space sound source that wish to deliver from indoor can be passed does quantification sound estimation method is proposing various kinds physical parameter to estimate degree of voice definition (Speech articulation) and reverberation. Result that evaluate sound source about MLS signal and Impluse signal by pistol in this treatise could know that converge in MLS and measurement error extent about reverberation time(RT) of sound benevolent person. And value is thought there is problem showing change irregularly about sound benevolent person of D50, C80 etc. Finally, in case is spread sound field in difference of sound pressure level, when measure about change of sound pressure level, sound benevolent person could know that there is no different effect.

  • PDF

A Real-time Implementation of G.729.1 Codec on an ARM Processor for the Improvement of VoWiFi Voice Quality (VoWiFi 음질 향상을 위한 G.729.1 광대역 코덱의 ARM 프로세서에의 실시간 구현)

  • Park, Nam-In;Kang, Jin-Ah;Kim, Hong-Kook
    • 한국HCI학회:학술대회논문집
    • /
    • 2008.02a
    • /
    • pp.230-235
    • /
    • 2008
  • This paper addresses issues associated with the real-time implementation of a wideband speech codec such as ITU-T G. 729. 1 on an ARM processor in order to provide an improved voice quality of a VoWiFi service. The real-time implementation features in optimizing the C-source code of G.729. 1 and replacing several parts of the codec algorithm with faster ones. The performance of the implementation is measured by the CPU time spent for G.729.1 on the ARM926EJ processor that is used for a VoWiFi phone. It is shown from the experiments that the G.729.1 codec works in real-time with better voice quality than G 729 codec that is conventionally used for VoIP or VoWiFi phones.

  • PDF